The present invention relates to the field of audio processing and more specifically to compensation for ambient noise in the listener's environment.
There are many systems for the application of filtering to noise suppression in an audio signal. In general, these inventions relate to the removal of noise present in a source signal from an origin or introduced into the signal through processing and transmission. Various forms of filtering may be applied which suppresses the noise signal in whole or in part, removing it from the source signal. Generally, these systems have adverse impacts upon the quality of the original signal. Further, these systems do not address noise in the environment of the listener, which cannot be filtered.
Conversely, some systems for the suppression of noise in the listener's environment also exist. These systems generally use noise cancellation to remove the disrupting external signal by adding sound projected through headphones which has the effect of countering the sound waves produced by the noise. In this case, the noise is completely canceled and the listener is generally unaware of the existence of the external noise—a result which can reduce the awareness of the listener to potential dangers in the environment.
In some prior art systems, dynamic volume compensation may be used to raise the volume of a source signal of interest over ambient background noise. However, these systems increase the gain in a spectrally uniform manner, raising the volume of all frequency components uniformly. This effect can distort the perception of music and speech due to the non-linear behavior of the human ear with respect to frequency and volume, and raise the volume to excessive levels.
Microphones and mechanical systems (e.g., computer software) can measure dBSPL, i.e., Decibels in sound power. A sound (e.g., 40 dBSPL) at a particular frequency (e.g., 1 kHz) sounds just as loud as the sound (e.g., 40 dBSPL) at a different frequency (e.g., 4 kHz) to a microphone or mechanical system. However, our hearing can be affected by the mechanical construction of our outer ear and/or slow variation in sensitivity across the basilar membrane due to fluid damping of the incident waves in the cochlear fluid. The variable sensitivity of human hearing is reflected in the Fletcher-Munson equal loudness contours and the equal-loudness contours from ISO 226:3003 revision. These contours show that perceived loudness varies according to the frequency and volume of the sound.
Since the human ear dynamically adjusts to sound intensity levels, the presence of background noise alters the threshold at which sounds begin to be perceived. As a result, ambient noise at a given frequency may make sounds at those frequencies that would otherwise be perceptible imperceptible in the presence of ambient noise. In order for the sound to be heard, it must be amplified over the background noise. The volume of the ambient noise therefore represents a degree of hearing impairment or baseline threshold elevation over which the sound must be amplified in order to be perceived.
This effect varies according to the spectral composition of the noise, that is, spectral components that are sufficiently far from the spectral composition of the noise will remain perceptible. Consequently, using the total intensity of the background noise to raise the intensity of the source uniformly will overly amplify bands which are not affected, possibly raising the volume to damaging levels. In order to amplify only those components which need compensation, the gains to the source signal must vary by spectral band, according to the spectral composition of the noise.
Moreover, due to the nonlinear response of the human ear, using the spectral intensity of the background noise at a particular band as the gain for the source at that band will produce excessive amplification. In order to compute the correct gain, a psychoacoustic model must be used to compute an appropriate gain for each frequency or band frequency. The psychoacoustic model is a mathematical representation of the dynamic behavior of the human ear, in terms of perceived loudness as a function of sound intensity. The intensity of the background noise as well as the source signal at a given frequency are inputs to this model, and the output is a desired gain for the source signal at that frequency or frequency band.
In the music industry, techniques such as parallel compression (commonly called New York Compression) have long been used to dynamically adjust the volume of quieter content in music in order to improve aesthetic qualities by bringing sub-threshold content above the hearing threshold. Parallel compression involves applying a linear gain to a signal which amplifies softer sounds and subsequently adding this amplified sound back in to the original signal. The result is generally a non-linear compression that amplifies softer tones without affecting the louder ones.
Parallel compression depends on a number of parameters including a threshold, which determines when the gain begins to fall off, as well as a compression ratio, and a makeup gain which adds an additional flat gain to match the final volume of the adjusted signal with the original signal. Usually these are fixed settings applied to a track, not dynamically adjusted over the course of the time. Parallel compression is also usually applied to a single signal band rather than used to perform multi-band compression.
The present invention features systems for dynamically adjusting audio signals by applying a gain to the signal in a spectrally varying manner to compensate for a noise source in the environment of the listener. The system obtains a threshold elevation for each frequency component by analyzing the spectral composition of the ambient noise. This threshold elevation is then used by a psychoacoustic model to determine an appropriate gain adjustment for the corresponding frequency component of the source signal. After applying the gains to the source signal, the system outputs the resulting signal to the speaker. The system allows a listener to hear the source signal over ambient noise, by applying a gain to the source that varies according to the spectral composition of the noise, rather than cancelling the noise, or applying a uniform volume adjustment to the source. The source is thus amplified without the removal of the noise signal, and without excessive volume increases. Systems may be incorporated into apparatuses including but not limited to mobile phones and music players.
The present invention utilizes parallel compression (New York compression) in its implementation, by dynamically altering the compression ratio, makeup gain, and threshold so as to approximate the compression curve determined according to the psychoacoustic model. In prior art uses of New York compression, these parameters are generally fixed throughout a track and dynamic adjustment of these parameters has not previously been conceived of to correct for ambient noise. Furthermore, use of parallel compression to approximate a function demanded by a psychoacoustic model is an entirely novel use of these techniques. The present invention thus applies existing techniques in a unique and novel way to create a unique and novel system for correcting for ambient noise.
Any feature or combination of features described herein are included within the scope of the present invention provided that the features included in any such combination are not mutually inconsistent as will be apparent from the context, this specification, and the knowledge of one of ordinary skill in the art. Additional advantages and aspects of the present invention are apparent in the following detailed description.